[Asterisk-Users] Little confused about Caller ID

Sun Jan 9 16:44:57 MST 2005

Caller Name is stored in a SCP.  It is a TCAP transaction.  The receiving
switch via SS7 recieves
the calling party number in the ISUP message of the SS7 datastream.  It is
normally in the IAM mesasge.  Then a TCAP CNAME query is launched from the
called switch thru
the STP's to a SCP which has the calling name database.  The TCAP query
returns back to the launching
switch the caller name.  LIDB is for operator services etc. CNAME is a TCAP
database lookup, much
like 800 number translations.

Tom C.

----- Original Message -----
From: "Alexander Lopez" <alex.lopez at opsys.com>
To: "C F" <shmaltz at gmail.com>; "Asterisk Users Mailing List - Non-Commercial
Discussion" <asterisk-users at lists.digium.com>
Sent: Sunday, January 09, 2005 5:30 PM
Subject: RE: [Asterisk-Users] Little confused about Caller ID

> OK here it goes..
>
> Caller ID is two parts or actually three:
>
> Part 1 Number only
> Part 2 Number + Name
> Part 3 Whole lotta stuff (also known as ADSI)
>
>
> Here is the US, I cannot speak for other countries.
>
> When party A places a call to Party B. Party A's Telco picks up the
> number, either from a table on the switch or passed from the PRI from
> Party A.  Then on the far side (Party B's Telco) the Telco does a lookup
> in the LIDB (Line Information Data Base) and associates a name with a
> number. This information is then passed as Part II CLID.
>
> I have simplified the process, leaving out many processes along the way
> but it should give some insight as to how the Name actually shows up on
> the other end.
>
> Most Telcos do not receive the Name as part of the data in the call
> through the tandems b/w Telcos, they opt rather to do the lookup in the
> LIDB themselves.
>
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of C F
> Sent: Sunday, January 09, 2005 6:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Little confused about Caller ID
>
> When calling to the PSTN (outside VOIP or *) then you will not be able
> to supply the name of callerID even if you have a PRI. The only thing
> you can provide is the number and the receiving switch of the call is
> the one responsibble for attaching a name to the phone number thru
> SS7. If you have a SS7 switch then you could in theory attach the name
> (I have never tried it, but that's what I was told).
> Hope this helps.
>
>
>
> On Sat, 8 Jan 2005 23:23:45 -0500 (EST), Samuel T. Cossette
> <digium at muel.org> wrote:
> > Hi,
> >
> > I've got the Caller ID name and number working with the application
> > SetCIDNumber and SetCIDName.
> >
> > [...]
> > exten => s,3,SetCIDNumber(4183289901)
> > exten => s,4,SetCIDName(Frank Black)
> > exten => s,5,Dial(IAX2/prov01/${DEST})
> > [...]
> >
> > You can also use SetCallerID(Frank Black <4183289901>), but no success
> for
> > me...
> >
> > bye,
> >
> > Samuel T. Cossette
> > samuel at levinux.org, 1.418.8o2.784o
> > << Well, that's for me to know and you to find out. >> Jeffrey, Blue
> Velvet
> >
> > > Hi Everybody,
> > >     Sure this has been covered a million times on wiki, but couldn't
> find
> > > an
> > > exact answer to my question.  I am using * to dial out to peoples
> phones
> > > to
> > > give them alerts of different things.  Problem is that the only
> Caller ID
> > > I
> > > can get working is the telephone number.  I am unable to display a
> name
> > > along with the number.  Thinking maybe its the phone receiving the
> call, I
> > > tried my cellphone and my house phone and I can only get the number
> to
> > > display.  If I leave the number portion out, Caller ID shows
> > > "Unavailable".
> > > Is there a simple way to get a Caller name setup?  I've tried
> examples on
> > > Wiki as well but I couldn't get them to work.
> > >
> > >
> > > ***** extensions.conf *********
> > > [general]
> > > static=yes
> > > writeprotect=no
> > >
> > > [globals]
> > > CONSOLE=Console/dsp ; Console interface for demo
> > > TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
> > >
> > > [sports]
> > > exten => s,1,ResponseTimeout,5
> > > exten => s,2,Answer
> > > exten => s,3,Wait(1)
> > > exten => s,4,Playback(sports/gafanaSports)
> > > exten => s,5,Goto(2000,2)
> > > exten => 2000,1,wait(1)
> > > exten => 2000,2,Background(sports/teams/theLosAngelesLakersU)
> > > exten => t,1,Playback(goodbye)
> > > exten => t,2,Hangup
> > >
> > > ******** sip.conf ***********
> > > [general]
> > > context=default
> > > port=5060
> > > srvlookup=yes
> > > allow=ulaw
> > > register => [id]:[pw]@[host]
> > > [gafana]
> > > type=peer
> > > secret=[secret]
> > > username=[username]
> > > host=[hostname]
> > >
> > >
> > > ******* test.call file **********
> > > Channel: SIP/[myNumber]@gafana
> > > CallerID: [My Number]
> > > MaxRetries: 0
> > > RetryTime: 300
> > > WaitTime: 45
> > > Context: sports
> > > Extension: s
> > > Priority: 1
> > >
> > >
> > > What do I need to add to be able to send a name as well and not just
> a
> > > number?
> > >
> > > Gabe
> > >
> > > _______________________________________________
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> >
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